[asterisk-users] [OpenSIPS-Users] 404 When BYE initiated by external callee

[Nathan Anderson]

On Tuesday, April 09, 2013 1:31 PM, Nick Khamis wrote:

> As I see asterisk rewrites the callid unexpectedly when initiating the
> INVITE with the SIP trunk (trace packet 4). 

[...]

> Asterisk has mapped the call with the two different ids together.

Nick,

As Joshua has already tried to explain to you, Asterisk is a B2BUA, not a SIP proxy.  This is by virtue of its nature and origins as a technology-agnostic PBX.  It is not "rewriting" the Call-ID.  It's generating an entirely new one because the INVITE that it generates is considered by Asterisk to be a competely new/separate call leg.  It then maintains a table of which call legs are "bridged" together into a single call, regardless of the underlying channel technology.  Because of how Asterisk works under-the-hood, it is also impossible for it to "pass on" Record-Route header fields to the other leg of the call.  It will, however, take appropriate action in passing any signalling events downstream (for example, in your case, a "BYE" will be sent to one call leg if it is received on the other, but NOT because it is proxying it; to boil it down, internally, the received "BYE" is translated to a generic "hang-up" event which the SIP channel driver takes and uses to generate a completely new "BYE" from scratch on the other leg).

I concur with Joshua: this is not an Asterisk problem, and what it is doing is completely reasonable.  I suspect that the "leads" you are chasing in this investigation will turn out to be a red herring.

-- 
Nathan Anderson
First Step Internet, LLC
nathana at fsr.com